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In: MissingRef
CLUE WG                                                          R. Even
Internet-Draft                                       Huawei Technologies
Intended status: Standards Track                               J. Lennox
Expires: August 31, 2017                                           Vidyo
                                                       February 27, 2017


               Mapping RTP streams to CLUE Media Captures
                   draft-ietf-clue-rtp-mapping-14.txt

Abstract

   This document describes how the Real Time transport Protocol (RTP) is
   used in the context of the CLUE protocol (ControLling mUltiple
   streams for tElepresence).  It also describes the mechanisms and
   recommended practice for mapping RTP media streams defined in Session
   Description Protocol (SDP) to CLUE Media Captures and defines a new
   RTP header extension (CaptureId).

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on August 31, 2017.

Copyright Notice

   Copyright (c) 2017 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  RTP topologies for CLUE . . . . . . . . . . . . . . . . . . .   3
   4.  Mapping CLUE Capture Encodings to RTP streams . . . . . . . .   4
   5.  MCC Constituent CaptureID definition  . . . . . . . . . . . .   5
     5.1.  RTCP CaptureID SDES Item  . . . . . . . . . . . . . . . .   5
     5.2.  RTP Header Extension  . . . . . . . . . . . . . . . . . .   6
   6.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .   6
   7.  Communication Security  . . . . . . . . . . . . . . . . . . .   7
   8.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .   8
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   10. Security Considerations . . . . . . . . . . . . . . . . . . .   8
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  10
     11.2.  Informative References . . . . . . . . . . . . . . . . .  11
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  13

1.  Introduction

   Telepresence systems can send and receive multiple media streams.
   The CLUE framework [I-D.ietf-clue-framework] defines Media Captures
   (MC) as a source of Media, from one or more Capture Devices.  A Media
   Capture may also be constructed from other Media streams.  A middle
   box can express conceptual Media Captures that it constructs from
   Media streams it receives.  A Multiple Content Capture (MCC) is a
   special Media Capture composed of multiple Media Captures.

   SIP Offer/Answer [RFC3264] uses SDP [RFC4566]  to describe the
   RTP[RFC3550] media streams.  Each RTP stream has a unique
   Synchronization Source (SSRC) within its RTP session.  The content of
   the RTP stream is created by an encoder in the endpoint.  This may be
   an original content from a camera or a content created by an
   intermediary device like an MCU (Multipoint Control Unit).

   This document makes recommendations for the CLUE architecture about
   how RTP and RTCP streams should be encoded and transmitted, and how
   their relation to CLUE Media Captures should be communicated.  The
   proposed solution supports multiple RTP topologies [RFC7667].

   With regards to the media (audio, video and timed text), systems that
   support CLUE use RTP for the media, SDP for codec and media transport
   negotiation (CLUE individual encodings) and the CLUE protocol for
   Media Capture description and selection.  In order to associate the



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   media in the different protocols there are three mapping that need to
   be specified:

   1.  CLUE individual encodings to SDP

   2.  RTP streams to SDP (this is not a CLUE specific mapping)

   3.  RTP streams to MC to map the received RTP steam to the current MC
       in the MCC.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC2119[RFC2119] and
   indicate requirement levels for RTP processing in compliant CLUE
   implementations.

   The definitions from the CLUE framework document
   [I-D.ietf-clue-framework] section 3 are used by this document as
   well.

3.  RTP topologies for CLUE

   The typical RTP topologies used by CLUE Telepresence systems specify
   different behaviors for RTP and RTCP distribution.  A number of RTP
   topologies are described in [RFC7667].  For CLUE telepresence, the
   relevant topologies include Point-to-Point, as well as Media-Mixing
   mixers, Media- Switching mixers, and Selective Forwarding Middleboxs.

   In the Point-to-Point topology, one peer communicates directly with a
   single peer over unicast.  There can be one or more RTP sessions,
   each sent on a separate 5-tuple, and having a separate SSRC space,
   with each RTP session carrying multiple RTP streams identified by
   their SSRC.  All SSRCs are recognized by the peers based on the
   information in the RTCP Source description (SDES) report that
   includes the CNAME and SSRC of the sent RTP streams.  There are
   different Point-to-Point use cases as specified in CLUE use case
   [RFC7205].  In some cases, a CLUE session which, at a high-level, is
   point-to-point may nonetheless have an RTP stream which is best
   described by one of the mixer topologies.  For example, a CLUE
   endpoint can produce composite or switched captures for use by a
   receiving system with fewer displays than the sender has cameras.
   The Media Capture may be described using an MCC.

   For the Media Mixer topology [RFC7667], the peers communicate only
   with the mixer.  The mixer provides mixed or composited media
   streams, using its own SSRC for the sent streams.  If needed by CLUE



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   endpoint, the conference roster information including conference
   participants, endpoints, media and media-id (SSRC) can be determined
   using the conference event package [RFC4575] element.

   Media-switching mixers and Selective Forwarding Middleboxes behave as
   described in [RFC7667]

4.  Mapping CLUE Capture Encodings to RTP streams

   The different topologies described in Section 3 create different SSRC
   distribution models and RTP stream multiplexing points.

   Most video conferencing systems today can separate multiple RTP
   sources by placing them into RTP sessions using the SDP description;
   the video conferencing application can also have some knowledge about
   the purpose of each RTP session.  For example, video conferencing
   applications that have a primary video source and a slides video
   source can send each media source in a separate RTP session with a
   content attribute [RFC4796] enabling different application behavior
   for each received RTP media source.  Demultiplexing is
   straightforward because each media capture is sent as a single RTP
   stream, with each RTP stream being sent in a separate RTP session, on
   a distinct UDP 5-tuple.  This will also be true for mapping the RTP
   streams to Media Captures Encodings if each Media Capture Encodings
   uses a separate RTP session, and the consumer can identify it based
   on the receiving RTP port.  In this case, SDP only needs to label the
   RTP session with an identifier that can be used to identify the Media
   Capture in the CLUE description.  The SDP label attribute serves as
   this identifier.

   Each Capture Encoding MUST be sent as a separate RTP stream.  CLUE
   endpoints MUST support sending each such RTP stream in a separate RTP
   session signalled by an SDP m= line.  They MAY also support sending
   some or all of the RTP streams in a single RTP session, using the
   mechanism described in [I-D.ietf-mmusic-sdp-bundle-negotiation] to
   relate RTP streams to SDP m= lines.

   MCCs bring another mapping issue, in that an MCC represents multiple
   Media Captures that can be sent as part of this MCC if configured by
   the consumer.  When receiving an RTP stream which is mapped to the
   MCC, the consumer needs to know which original MC it is in order to
   get the MC parameters from the advertisement.  If a consumer
   requested a MCC, the original MC does not have a capture encoding, so
   it cannot be associated with an m-line using a label as described in
   CLUE signaling [I-D.ietf-clue-signaling].  This is important, for
   example, to get correct scaling information for the original MC,
   which may be different for the various MCs that are contributing to
   the MCC.



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5.  MCC Constituent CaptureID definition

   For a MCC which can represent multiple switched MCs there is a need
   to know which MC is represented in the current RTP stream at any
   given time.  This requires a mapping from the SSRC of the RTP stream
   conveying a particular MCC to the constituent MC.  In order to
   address this mapping this document defines an RTP header extension
   and SDES item that includes the captureID of the original MC,
   allowing the consumer to use the original source MC's attributes like
   the spatial information.

   This mapping temporarily associates the SSRC of the RTP stream
   conveying a particular MCC with the captureID of the single original
   MC that is currently switched into the MCC.  This mapping cannot be
   used for the composed case where more than one original MC is
   composed into the MCC simultaneously.

   If there is only one MC in the MCC then the media provider MUST send
   the captureID of the current constituent MC in the RTP Header
   Extension and as a RTCP CaptureID SDES item.  When the media provider
   switches the MC it sends within an MCC, it MUST send the captureID
   value for the MC just switched into the MCC in an RTP Header
   Extension and as a RTCP CaptureID SDES item as specified in [RFC7941]

   If there is more than one MC composed into the MCC then the media
   provider MUST NOT send any of the MCs' captureIDs using this
   mechanism.  However, if an MCC is sending contributing source (CSRC)
   information in the RTP header for a composed capture, it MAY send the
   captureID values in the RTCP SDES packets giving source information
   for the SSRC values sent as contributing sources (CSRCs).

   If the media provider sends the captureID of a single MC switched
   into an MCC, then later sends one composed stream of multiple MCs in
   the same MCC, it MUST send the special value "-", a single dash
   character, as the captureID RTP Header Extension and RTCP CaptureID
   SDES item.  The single dash character indicates there is no
   applicable value for the MCC constituent CaptureID.  The media
   consumer interprets this as meaning that any previous CaptureID value
   associated with this SSRC no longer applies.  As
   [I-D.ietf-clue-data-model-schema] defines the captureID syntax as
   "xs:ID", the single dash character is not a legal captureID value, so
   there is no possibility of confusing it with an actual captureID.

5.1.  RTCP CaptureID SDES Item

   This document specifies a new RTCP SDES item.





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    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   CaptId=TBA  |     length    | CaptureID                     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   ....        |
   +-+-+-+-+-+-+-+-+

   Note to the RFC Editor: Please replace TBA with the value assigned by
   IANA.

   This CaptureID is a variable-length UTF-8 string corresponding either
   to a CaptureID negotiated in the CLUE protocol, or the single
   character "-".

   This SDES item MUST be sent in an SDES packet within a compound RTCP
   packet unless support for Reduced-size RTCP has been negotiated as
   specified in RFC 5506 [RFC5506], in which case it can be sent as an
   SDES packet in a non-compound RTCP packet.

5.2.  RTP Header Extension

   The CaptureID is also carried in an RTP header extension [RFC5285],
   using the mechanism defined in [RFC7941].

   Support is negotiated within SDP using the URN "urn:ietf:params:rtp-
   hdrext:sdes:CaptureID".

   The CaptureID is sent in a RTP Header Extension because for switched
   captures, receivers need to know which original MC corresponds to the
   media being sent for an MCC, in order to correctly apply geometric
   adjustments to the received media.

   As discussed in [RFC7941], there is no need to send the CaptId Header
   Extension with all RTP packets.  Senders MAY choose to send it only
   when a new MC is sent.  If such a mode is being used, the header
   extension SHOULD be sent in the first few RTP packets to reduce the
   risk of losing it due to packet loss.  See [RFC7941] for more
   discussion of this.

6.  Examples

   In this partial advertisement the Media Provider advertises a
   composed capture VC7 made of a big picture representing the current
   speaker (VC3) and two picture-in-picture boxes representing the
   previous speakers (the previous one -VC5- and the oldest one -VC6).





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<ns2:mediaCapture xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
      xsi:type="ns2:videoCaptureType" captureID="VC7" mediaType="video">
          <ns2:captureSceneIDREF>CS1</ns2:captureSceneIDREF>
          <ns2:nonSpatiallyDefinable>true</ns2:nonSpatiallyDefinable>
          <ns2:content>
                <ns2:captureIDREF>VC3</ns2:captureIDREF>
                <ns2:captureIDREF>VC5</ns2:captureIDREF>
                <ns2:captureIDREF>VC6</ns2:captureIDREF>
          </ns2:content>
                  <ns2:maxCaptures>3</ns2:maxCaptures>
            <ns2:allowSubsetChoice>false</ns2:allowSubsetChoice>
          <ns2:description lang="en">big picture of the current speaker
            pips about previous speakers</ns2:description>
            <ns2:priority>1</ns2:priority>
            <ns2:lang>it</ns2:lang>
            <ns2:mobility>static</ns2:mobility>
            <ns2:view>individual</ns2:view>
        </ns2:mediaCapture>


   In this case the media provider will send capture IDs VC3, VC5 or VC6
   as an RTP header extension and RTCP SDES message for the RTP stream
   associated with the MC.

   Note that this is part of the full advertisement message example from
   CLUE data model[I-D.ietf-clue-data-model-schema] example and is not a
   valid xml document.

7.  Communication Security

   CLUE endpoints MUST support RTP/SAVPF profile and SRTP [RFC3711].
   CLUE endpoints MUST support DTLS [RFC6347] and DTLS-SRTP [RFC5763]
   [RFC5764] for SRTP keying.

   All media channels SHOULD be secure via SRTP and the RTP/SAVPF
   profile unless the RTP media and its associated RTCP are secure by
   other means (see [RFC7201] [RFC7202]).

   All CLUE implementations MUST implement DTLS 1.0, with the cipher
   suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA with the the P-256 curve
   [FIPS186].  The DTLS-SRTP protection profile
   SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.Encrypted SRTP
   Header extensions [RFC6904] MUST be supported.

   Implementations SHOULD implement DTLS 1.2 with the
   TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.
   Implementations MUST favor cipher suites which support PFS over non-
   PFS cipher suites and SHOULD favor AEAD over non-AEAD cipher suites.



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   NULL Protection profiles MUST NOT be used for RTP or RTCP.

   CLUE endpoint MUST generate short-term persistent RTCP CNAMES, as
   specified in [RFC7022], and thus can't be used for long term tracking
   of the users.

8.  Acknowledgments

   The authors would like to thanks Allyn Romanow and Paul Witty for
   contributing text to this work.  Magnus Westerlund helped drafting
   the security section.

9.  IANA Considerations

   This document defines a new extension URI in the RTP SDES Compact
   Header Extensions subregistry of the Real-Time Transport Protocol
   (RTP) Parameters registry, according to the following data:

      Extension URI: urn:ietf:params:rtp-hdrext:sdes:CaptId

      Description: CLUE CaptId

      Contact: ron.even.tlv@gmail.com

      Reference: RFC XXXX

   The IANA is requested to register one new RTCP SDES items in the
   "RTCP SDES Item Types" registry, as follows:

      Value    Abbrev        Name                         Reference
         TBA      CCID           CLUE CaptId          [RFCXXXX]

   Note to the RFC Editor: Please replace RFCXXXX with this RFC number.

10.  Security Considerations

   The security considerations of the RTP specification, the RTP/SAVPF
   profile, and the various RTP/RTCP extensions and RTP payload formats
   that form the complete protocol suite described in this memo apply.
   It is not believed there are any new security considerations
   resulting from the combination of these various protocol extensions.

   The Extended Secure RTP Profile for Real-time Transport Control
   Protocol (RTCP)-Based Feedback [RFC5124] (RTP/SAVPF) provides
   handling of fundamental issues by offering confidentiality, integrity
   and partial source authentication.  A mandatory to implement and use
   media security solution is created by combining this secured RTP




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   profile and DTLS-SRTP keying [RFC5764] as defined in the
   communication security section of this memo Section 7

   RTCP packets convey a Canonical Name (CNAME) identifier that is used
   to associate RTP packet streams that need to be synchronised across
   related RTP sessions.  Inappropriate choice of CNAME values can be a
   privacy concern, since long-term persistent CNAME identifiers can be
   used to track users across multiple calls.  The communication
   security section of this memo Section 7 mandates generation of short-
   term persistent RTCP CNAMES, as specified in [RFC7022] so they can't
   be used for long term tracking of the users.

   Some potential denial of service attacks exist if the RTCP reporting
   interval is configured to an inappropriate value.  This could be done
   by configuring the RTCP bandwidth fraction to an excessively large or
   small value using the SDP "b=RR:" or "b=RS:" lines [RFC3556], or some
   similar mechanism, or by choosing an excessively large or small value
   for the RTP/AVPF minimal receiver report interval (if using SDP, this
   is the "a=rtcp-fb:... trr-int" parameter) [RFC4585]  The risks are as
   follows:

   1.  the RTCP bandwidth could be configured to make the regular
       reporting interval so large that effective congestion control
       cannot be maintained, potentially leading to denial of service
       due to congestion caused by the media traffic;

   2.  the RTCP interval could be configured to a very small value,
       causing endpoints to generate high rate RTCP traffic, potentially
       leading to denial of service due to the non-congestion controlled
       RTCP traffic; and

   3.  RTCP parameters could be configured differently for each
       endpoint, with some of the endpoints using a large reporting
       interval and some using a smaller interval, leading to denial of
       service due to premature participant timeouts due to mismatched
       timeout periods which are based on the reporting interval (this
       is a particular concern if endpoints use a small but non-zero
       value for the RTP/AVPF minimal receiver report interval (trr-int)
       [RFC4585], as discussed in [I-D.ietf-avtcore-rtp-multi-stream]).

   Premature participant timeout can be avoided by using the fixed (non-
   reduced) minimum interval when calculating the participant timeout
   ([I-D.ietf-avtcore-rtp-multi-stream]).  To address the other
   concerns, endpoints SHOULD ignore parameters that configure the RTCP
   reporting interval to be significantly longer than the default five
   second interval specified in [RFC3550] (unless the media data rate is
   so low that the longer reporting interval roughly corresponds to 5%
   of the media data rate), or that configure the RTCP reporting



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   interval small enough that the RTCP bandwidth would exceed the media
   bandwidth.

   The guidelines in [RFC6562] apply when using variable bit rate (VBR)
   audio codecs such as Opus.

   The use of the encryption of the header extensions are RECOMMENDED,
   unless there are known reasons, like RTP middleboxes performing voice
   activity based source selection or third party monitoring that will
   greatly benefit from the information, and this has been expressed
   using API or signalling.  If further evidence are produced to show
   that information leakage is significant from audio level indications,
   then use of encryption needs to be mandated at that time.

   In multi-party communication scenarios using RTP Middleboxes; this
   middleboxes are REQUIRED, by this protocol, to not weaken the
   sessions' security.  The middlebox SHOULD maintain the
   confidentiality, integrity and perform source authentication.  The
   middlebox MAY perform checks that prevents any endpoint participating
   in a conference to impersonate another.  Some additional security
   considerations regarding multi-party topologies can be found in
   [RFC7667]

   The CaptureID is created as part of the CLUE protocol.  The CaptId
   SDES item is used to convey the same CaptureID value in the SDES
   item.  When sending the SDES item the security consideration
   specified in the security section of [RFC7941] and in the
   communication security section of this memo Section 7 are applicable.
   Note that since the CaptureID is carried also in CLUE protocol
   messages it is RECOMMENDED that this SDES item use at least similar
   protection profiles as the CLUE protocol messages carried in the CLUE
   data channel.  .

11.  References

11.1.  Normative References

   [I-D.ietf-clue-data-model-schema]
              Presta, R. and S. Romano, "An XML Schema for the CLUE data
              model", draft-ietf-clue-data-model-schema-17 (work in
              progress), August 2016.

   [I-D.ietf-clue-framework]
              Duckworth, M., Pepperell, A., and S. Wenger, "Framework
              for Telepresence Multi-Streams", draft-ietf-clue-
              framework-25 (work in progress), January 2016.





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   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-36 (work in progress), October 2016.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, DOI 10.17487/RFC3711, March 2004,
              <http://www.rfc-editor.org/info/rfc3711>.

   [RFC5763]  Fischl, J., Tschofenig, H., and E. Rescorla, "Framework
              for Establishing a Secure Real-time Transport Protocol
              (SRTP) Security Context Using Datagram Transport Layer
              Security (DTLS)", RFC 5763, DOI 10.17487/RFC5763, May
              2010, <http://www.rfc-editor.org/info/rfc5763>.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764,
              DOI 10.17487/RFC5764, May 2010,
              <http://www.rfc-editor.org/info/rfc5764>.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <http://www.rfc-editor.org/info/rfc6347>.

   [RFC6904]  Lennox, J., "Encryption of Header Extensions in the Secure
              Real-time Transport Protocol (SRTP)", RFC 6904,
              DOI 10.17487/RFC6904, April 2013,
              <http://www.rfc-editor.org/info/rfc6904>.

   [RFC7941]  Westerlund, M., Burman, B., Even, R., and M. Zanaty, "RTP
              Header Extension for the RTP Control Protocol (RTCP)
              Source Description Items", RFC 7941, DOI 10.17487/RFC7941,
              August 2016, <http://www.rfc-editor.org/info/rfc7941>.

11.2.  Informative References

   [FIPS186]  National Institute of Standards and Technology, "Digital
              Signature Standard", FIPS PUB 186-4, July 2013.





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   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-11 (work in progress),
              December 2015.

   [I-D.ietf-clue-signaling]
              Kyzivat, P., Xiao, L., Groves, C., and R. Hansen, "CLUE
              Signaling", draft-ietf-clue-signaling-10 (work in
              progress), January 2017.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              DOI 10.17487/RFC3264, June 2002,
              <http://www.rfc-editor.org/info/rfc3264>.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3556]  Casner, S., "Session Description Protocol (SDP) Bandwidth
              Modifiers for RTP Control Protocol (RTCP) Bandwidth",
              RFC 3556, DOI 10.17487/RFC3556, July 2003,
              <http://www.rfc-editor.org/info/rfc3556>.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC4575]  Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
              Session Initiation Protocol (SIP) Event Package for
              Conference State", RFC 4575, DOI 10.17487/RFC4575, August
              2006, <http://www.rfc-editor.org/info/rfc4575>.

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,
              <http://www.rfc-editor.org/info/rfc4585>.

   [RFC4796]  Hautakorpi, J. and G. Camarillo, "The Session Description
              Protocol (SDP) Content Attribute", RFC 4796,
              DOI 10.17487/RFC4796, February 2007,
              <http://www.rfc-editor.org/info/rfc4796>.






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Internet-Draft             RTP mapping to CLUE             February 2017


   [RFC5124]  Ott, J. and E. Carrara, "Extended Secure RTP Profile for
              Real-time Transport Control Protocol (RTCP)-Based Feedback
              (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February
              2008, <http://www.rfc-editor.org/info/rfc5124>.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July
              2008, <http://www.rfc-editor.org/info/rfc5285>.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
              2009, <http://www.rfc-editor.org/info/rfc5506>.

   [RFC6562]  Perkins, C. and JM. Valin, "Guidelines for the Use of
              Variable Bit Rate Audio with Secure RTP", RFC 6562,
              DOI 10.17487/RFC6562, March 2012,
              <http://www.rfc-editor.org/info/rfc6562>.

   [RFC7022]  Begen, A., Perkins, C., Wing, D., and E. Rescorla,
              "Guidelines for Choosing RTP Control Protocol (RTCP)
              Canonical Names (CNAMEs)", RFC 7022, DOI 10.17487/RFC7022,
              September 2013, <http://www.rfc-editor.org/info/rfc7022>.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
              <http://www.rfc-editor.org/info/rfc7201>.

   [RFC7202]  Perkins, C. and M. Westerlund, "Securing the RTP
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
              2014, <http://www.rfc-editor.org/info/rfc7202>.

   [RFC7205]  Romanow, A., Botzko, S., Duckworth, M., and R. Even, Ed.,
              "Use Cases for Telepresence Multistreams", RFC 7205,
              DOI 10.17487/RFC7205, April 2014,
              <http://www.rfc-editor.org/info/rfc7205>.

   [RFC7667]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667,
              DOI 10.17487/RFC7667, November 2015,
              <http://www.rfc-editor.org/info/rfc7667>.

Authors' Addresses








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Internet-Draft             RTP mapping to CLUE             February 2017


   Roni Even
   Huawei Technologies
   Tel Aviv
   Israel

   Email: roni.even@huawei.com


   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack, NJ  07601
   US

   Email: jonathan@vidyo.com



































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